Prepare for splitting FrameType into AudioFrameType and VideoFrameType
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index e5a7684..7667b71 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -103,7 +103,7 @@
return num_10ms_frames;
}
- int SendData(FrameType frame_type,
+ int SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
@@ -139,7 +139,7 @@
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.
uint32_t last_packet_send_timestamp_;
- FrameType last_frame_type_;
+ AudioFrameType last_frame_type_;
};
#if defined(WEBRTC_ANDROID)
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index b6110b6..4c34e41 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -123,7 +123,7 @@
// This method receives the callback from ACM when a new packet is produced.
int32_t AcmSendTestOldApi::SendData(
- FrameType frame_type,
+ AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
diff --git a/modules/audio_coding/acm2/acm_send_test.h b/modules/audio_coding/acm2/acm_send_test.h
index 24d230b..744d015 100644
--- a/modules/audio_coding/acm2/acm_send_test.h
+++ b/modules/audio_coding/acm2/acm_send_test.h
@@ -50,7 +50,7 @@
std::unique_ptr<Packet> NextPacket() override;
// Inherited from AudioPacketizationCallback.
- int32_t SendData(FrameType frame_type,
+ int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
@@ -75,7 +75,7 @@
bool codec_registered_;
int test_duration_ms_;
// The following member variables are set whenever SendData() is called.
- FrameType frame_type_;
+ AudioFrameType frame_type_;
int payload_type_;
uint32_t timestamp_;
uint16_t sequence_number_;
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 1547b37..a4b64b1 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -393,7 +393,7 @@
RTPFragmentationHeader my_fragmentation;
ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
- FrameType frame_type;
+ AudioFrameType frame_type;
if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
frame_type = kEmptyFrame;
encoded_info.payload_type = previous_pltype;
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 4ee9add..797b9b1 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -104,7 +104,7 @@
last_payload_type_(-1),
last_timestamp_(0) {}
- int32_t SendData(FrameType frame_type,
+ int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
@@ -129,7 +129,7 @@
return rtc::checked_cast<int>(last_payload_vec_.size());
}
- FrameType last_frame_type() const {
+ AudioFrameType last_frame_type() const {
rtc::CritScope lock(&crit_sect_);
return last_frame_type_;
}
@@ -151,7 +151,7 @@
private:
int num_calls_ RTC_GUARDED_BY(crit_sect_);
- FrameType last_frame_type_ RTC_GUARDED_BY(crit_sect_);
+ AudioFrameType last_frame_type_ RTC_GUARDED_BY(crit_sect_);
int last_payload_type_ RTC_GUARDED_BY(crit_sect_);
uint32_t last_timestamp_ RTC_GUARDED_BY(crit_sect_);
std::vector<uint8_t> last_payload_vec_ RTC_GUARDED_BY(crit_sect_);
@@ -430,7 +430,7 @@
// that is contain comfort noise.
const struct {
int ix;
- FrameType type;
+ AudioFrameType type;
} expectation[] = {
{2, kAudioFrameCN}, {5, kEmptyFrame}, {8, kEmptyFrame},
{11, kAudioFrameCN}, {14, kEmptyFrame}, {17, kEmptyFrame},
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 7e5bf1b..0621473 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -40,7 +40,7 @@
public:
virtual ~AudioPacketizationCallback() {}
- virtual int32_t SendData(FrameType frame_type,
+ virtual int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
@@ -53,7 +53,7 @@
public:
virtual ~ACMVADCallback() {}
- virtual int32_t InFrameType(FrameType frame_type) = 0;
+ virtual int32_t InFrameType(AudioFrameType frame_type) = 0;
};
class AudioCodingModule {
diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc
index 14c6e58..443dfd8 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -107,7 +107,7 @@
ssrc_(ssrc),
timestamp_rate_hz_(timestamp_rate_hz) {}
- int32_t SendData(FrameType frame_type,
+ int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index adfc0d5..d54faa7 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -18,7 +18,7 @@
namespace webrtc {
-int32_t Channel::SendData(FrameType frameType,
+int32_t Channel::SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index 4d7f0b7..6a55b06 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -47,7 +47,7 @@
Channel(int16_t chID = -1);
~Channel() override;
- int32_t SendData(FrameType frameType,
+ int32_t SendData(AudioFrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 28ee8aa..c961fe5 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -33,8 +33,10 @@
}
int32_t TestPacketization::SendData(
- const FrameType /* frameType */, const uint8_t payloadType,
- const uint32_t timeStamp, const uint8_t* payloadData,
+ const AudioFrameType /* frameType */,
+ const uint8_t payloadType,
+ const uint32_t timeStamp,
+ const uint8_t* payloadData,
const size_t payloadSize,
const RTPFragmentationHeader* /* fragmentation */) {
_rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index cdfc706..6dc7bc9 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -28,7 +28,7 @@
public:
TestPacketization(RTPStream *rtpStream, uint16_t frequency);
~TestPacketization();
- int32_t SendData(const FrameType frameType,
+ int32_t SendData(const AudioFrameType frameType,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index 81b83c0..52518ac 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -60,7 +60,7 @@
return;
}
-int32_t TestPack::SendData(FrameType frame_type,
+int32_t TestPack::SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
diff --git a/modules/audio_coding/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h
index 3125efe..d8a7711 100644
--- a/modules/audio_coding/test/TestAllCodecs.h
+++ b/modules/audio_coding/test/TestAllCodecs.h
@@ -25,7 +25,7 @@
void RegisterReceiverACM(AudioCodingModule* acm);
- int32_t SendData(FrameType frame_type,
+ int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 2c71f46..2fa56de 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -40,7 +40,7 @@
return;
}
-int32_t TestPackStereo::SendData(const FrameType frame_type,
+int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
const uint8_t payload_type,
const uint32_t timestamp,
const uint8_t* payload_data,
diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
index da10bf1..9a44a10 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -31,7 +31,7 @@
void RegisterReceiverACM(AudioCodingModule* acm);
- int32_t SendData(const FrameType frame_type,
+ int32_t SendData(const AudioFrameType frame_type,
const uint8_t payload_type,
const uint32_t timestamp,
const uint8_t* payload_data,
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index 7c04b22..b22e97e 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -33,7 +33,7 @@
ResetStatistics();
}
-int32_t ActivityMonitor::InFrameType(FrameType frame_type) {
+int32_t ActivityMonitor::InFrameType(AudioFrameType frame_type) {
counter_[frame_type]++;
return 0;
}
diff --git a/modules/audio_coding/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h
index f2358e7..36d5f95 100644
--- a/modules/audio_coding/test/TestVADDTX.h
+++ b/modules/audio_coding/test/TestVADDTX.h
@@ -25,7 +25,7 @@
class ActivityMonitor : public ACMVADCallback {
public:
ActivityMonitor();
- int32_t InFrameType(FrameType frame_type);
+ int32_t InFrameType(AudioFrameType frame_type);
void PrintStatistics();
void ResetStatistics();
void GetStatistics(uint32_t* stats);